I don't know what's the reason to not have foobar2000 not have a function to undo tag.
That's easy, most people don't have issues with the problem you're describing and/or have preventive measures in place to help with such problems (backups). This seems to be a niche fringe request imho and considering the response in this thread you shouldn't expect anyone to jump in and make something and dedicate their time to it.
Last post by fohrums -
I'm surprised this hasn't been brought up enough. I would love this feature too even if that means a component based method of doing-so. I'm with wcs13 when dealing with many tracks it's impossible to not make a mistake. Whether the mistake was you thought the select-all or CTRL-A function for the current playlist is the album list you wanted to re-edit or it was your 50,000 track library and all of them now have the same Artist Name = YIKES!
I don't know what's the reason to not have foobar2000 not have a function to undo tag changes especially with a great history support for that matter, too. Again, Foobar2000 needs this safety function feature and not more anymore aesthetic ones.
Last post by rutra80 -
Nothing is perfect, FLAC predictors aren't either. There surely exist data patterns that you can add to sound file, which don't add actual meaningful information, but will cause FLAC files to bloat because its predictors won't handle them efficiently. FLAC isn't a format that aims at maximum compression.
Last post by kaiser.lima -
My experience in upload to youtube shows me that if you use V1 HE-AAC, or >190 LE-AAC, for example, youtube will convert to 120k LE-AAC and its crops the sound above 16kHz . The best result I observed in streams sites was upload mp3lame VBR (0) audio streams at 48100 Hz encoded inside the H.264 video, them the server choose the better compress method, like done in .wav files. Some youtube accounts appear have special permissions that allows the 190k AAC ( mp4a.40.2@192k), but its apear to be automatic based in popularity (like the VEVO i.e.). The mp3lame VBR (0) at least "brings" the sound until 20 kHz to youtube server, i think, in "sane" internet bandwidth . HE-AAC appear to have problems in youtube because the codec licences related at this format. LE-AAC are 99% converted to poor 120k.
Last post by kaiser.lima -
I use ffmpeg with aac_fdk from Zeranoe's FFmpeg Builds Home Page: <http://ffmpeg.zeranoe.com/builds/> I found better result with this bat command, bring the files you want to the flie .bat you create, the files wil place C:Temp in this example.
------------------ @ECHO ON
FOR %%A IN (%*) DO ( "C:\Program Files\ffmpeg-20151003-git-061b67f-win32-shared\bin\ffmpeg" -i %%A -vn -c:a libfdk_aac -profile:a aac_he -ar 88200 -b:a 160k -y "C:\Temp\%%~nA.m4a" ) --------- -The result is a 160 kbps he-aac file that sounds equivalent at 320 kbps lame mp3 (saved at C:\Temp in this example) -if the original file have 48 kHz the result are yet better, you can use a video file i. e. -Original studio editions (masters) when done at 60, 88 kHz have good results yet in 20-30 KHz audio spectrum! (observed in Adobe Audition audiograms and some mastering i done). Good to producers! -the "ffmpeg-20151003-git-061b67f-win32-shared" part must be the folder name of bin/ffmpeg.exe that you have after download and place the program in "Program Files" folder, it may change in your version! -some audio systems (or old tv media boxes) can't handle the HE part or files at 88,2 kHz, if you have them, sell it!!
If you're fine with that bitrate you may as well just use a time domain subband codec like MusePack or hell, even MP2, and enjoy the perfect temporal resolution. Throwing more bits at a transform codec doesn't do much to fix their fundamental shortcomings beyond 192 kbps.
Still does not change the fact that i love to convert with Opus - it has become my favorite Open Source codec. Flawless in every kind of ways
Last post by tedsmith -
When in foo_input_sacd's Output Mode of "DSD+PCM" you can set the "PCM Samplerate"to any of the options just fine. They will all work essentially identically for most systems. They don't affect the bits going to the DAC so don't worry about it. (They matter more when you convert DSD to PCM for your DAC...) Similarly the DSD2PCM can be any filter you like when you are using "DSD+PCM".
From what I can understand, by putting output in DSD + PCM, Foobar will send my DAC a pure DSD stream when playing DSD files, but convert a PCM version to use internally for visualizations. But then I have the option of selecting a samplerate which doesn't align with with 192 without downconversion. I have 44.1, 88.2, 176.4, and 352.8.
I'm trying to set up playback of hi-res files using Foobar2000, but I'm having a really hard time figuring out what all the settings mean, and what options to choose.
I have a Sony CAS-1 DAC/amp, which is powering a pair of LS50's. It supports DSD, FLAC, and PCM up to 32-bit/192khz. Right now the best files I have are DSD 64/128 and WAV 24-bit/192khz. I have a couple 32-bit WAV files too, but they're 48khz. Unfortunately the CAS-1 doesn't support DSD playback beyond 2.8Mhz, so my DSD 128 files can't be played.
Here's a list of issues I've tried to research on my own but can't find answers to:
In Windows, I have the USB audio output of my Surface Pro set to 32-bit/192khz. Is that the right setting to be in? When I run foobar, will it automatically change the output based on the track, or is everything going to get converted to 32/192?
In Foobar, the default maximum sample rate is 88.2khz. Will I affect anything by changing it to 192khz?
I downloaded and installed a few Foobar components to support DSD playback. I have ASIO support, DSD Processor, DSDIFF Decoder, and Super Audio CD Decoder installed into Foobar. Are any of those redundant/unnecessary? I've tried playing around but can't figure it out.
Under preferences>playback>output, I have my output device set to "DSD: ASIO : Sony Audio Driver". I also have the choices of "ASIO: DSD Transcoder (DoP/Native)", "ASIO: Sony Audio Driver", "DS: Primary Audio Driver", and "DS: CAS-1 (Sony Audio)". What do all those settings mean, and do I have it in the right one? I don't want to click through them all and test because last time I tried to, it screwed with playback and I started getting error messages, no matter output I chose after.
Under preferences>playback>output, I have to choose between 8, 16, 24, and 32-bit for output data format. I have it set to 32-bit. Will that make Foobar automatically upconvert anything that's not already 32-bit, and if so does that affect sound quality? Also, does this affect DSD files?
Under preferences>tools, there's a section called "DSD processor" with a box labeled "Use DSD Processor" that I can select. When I select it, nothing changes. What does it do?
Under preferences>tools>SACD, there's an option labeled "output mode". The choices are "PCM", "DSD", and "PCM+DSD". What do each of these do, and what's the best one? Online I found instructions saying to choose "DSD", but does that convert everything to DSD? Does PCM+DSD allow for both codecs to be transmitted without any kind of conversion between the two?
Last question: under that same SACD section, there's another option labeled "DSD processor" with the options "DSD processor". and "none". What does this do?
This is especially frustrating because I have a background working in both tech and audio. If I don't know the answer to something, I know how to find it online. But this has got me beat. I just want my music getting to the DAC as untouched as possible. Thanks everyone.